Hello all, Hoping someone can help me with an issue...I have i .call file which calls out on a SIP channel and connects to an extension which dials another SIP channel. (both via voip providers) - both to PSTN.
Problem is, hanging up the POTS phone doesn't release the channel (either one - hanging up the calling channel or the destination doesn't do it). Using IAX instead of SIP works better; it releases the voip channels but not the POTS channels (i.e. the POTS phones don't immediately go back to a dial tone or fast busy). I'm using asterisk 1.4 here are the relevant bits: the extension: exten=> 157,1,Answer exten=> 157,2,Dial(IAX2/[EMAIL PROTECTED]/PSTNnumberToCall2, 60) exten=> 157,3,Hangup the .call file: # Create the call on group 2 dial lines and set up # some re-try timers # Channel: IAX2/[EMAIL PROTECTED]/PSTNnumberToCall1 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Extension: 157 Priority: 1 if anyone can shed some light on this i'd be eternally grateful, first of all, why if i glue 2 SIP channels together hanging up the POTS phone doesn't release the SIP channels, and second why if i glue two IAX channels together it doesnt release the POTS lines. thanks, yair
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